In the past, mobile communication operators of one country or region only deploy access networks of one standard. Under this network condition, a user enjoys services through a single-mode terminal. Taking a worldwide view, networks of different standards coexist. With the selections of the operators in deploying networks as well as the mergers and acquisitions between the operators, one operator may operate networks of multiple standards at the same time. On the other hand, the growing demand of the user for mobile applications greatly promotes the rapid development of access technologies, so that many new access technologies, for example, Wireless Fidelity, WiFi, Worldwide Interoperability for Microwave Access, WiMAX, etc., come forth continuously. Therefore, providing the users with a seamless service handover to ensure service continuity under heterogeneous access technologies has become an urgent need for the operators to satisfy the users' requirements and enhance their own competitiveness. Currently, the use of a same set of core networks to support different access technologies and meanwhile support a terminal of different access modes (i.e., a multi-mode terminal, and a CSI terminal and a VCC terminal are both applications of the multi-mode terminal) is put forward to satisfy this need.
The combination of circuit-switched, CS, and Internet protocol, IP, multimedia system, IMS, services (CSI, combining CS bearers with IMS) is proposed by the 3rd Generation Project Partnership, 3GPP, for combining advantages of a CS domain and an IMS domain by enhancing terminal capabilities and providing the users with multimedia service experience in conjunction with an access network and an IMS network entity CSI-application server, CSI-AS, in the circumstance that the access network side supporting a packet-switched, PS, domain cannot bear real-time media. That is, a real-time media component, for example, audio, video, etc., is borne by a call in a CS network, and a non-real-time media component, for example, text, etc., is borne by a session on a packet-based network, and the video may also be borne in the PS domain if it is bearable to the user. Meanwhile, in order to ensure the service experience, the CSI requires that the CS and IMS session must be terminated to the same terminal of a peer end user, and the terminal of the user associates the CS call and the IMS session and provides them to an end user. A combinational service may be generated by adding a CS voice or a multimedia call in the IMS session or adding several IMS sessions to an existing CS voice or multimedia call.
In the above CSI, the CSI-AS has the following functions: (1) selection of whether to combine sessions established on different access networks initiated by the CSI user (when calling) according to network policies; (2) splitting of a received multimedia session to different access networks registered by the CSI terminal for connection (when called); (3) generation of charging information related to the CSI; and (4) supplementary service processing related to the CSI.
In the above CSI, in terms of access network capabilities, in order to associate sessions by using different access technologies at the same time, the CSI requires a GSM/EDGE radio access network, GERAN, network to support the dual transfer mode, DTM, technology, or requires a UMTS terrestrial radio access network, UTRAN, network to support the multi radio access bearer, multi RAB, technology. Furthermore, if it is not limited to the combination between the CS real-time media call and the IMS non-real-time media session, the terminal may also provide an IMS session combination on networks of different access technologies to the user, and such terminal belongs to the CSI terminal in general sense. For example, a session combination between an IMS voice call on a wireless local area network, WLAN, bearer and an IMS text of a PS bearer is provided to the user.
Taking the splitting of the multimedia IMS session by the CSI-AS as an example, referring to FIG. 1, after receiving the multimedia IMS session, the CSI-AS splits the IMS session into a CS call and an IMS call, i.e., bears a real-time media component, for example, audio, video, etc., on a CS network, and bears a non-real-time media component, for example, text, etc., on a packet-based network (the video may also be borne in the PS domain if it is bearable to the user). Meanwhile, in order to ensure the service experience, the CSI requires that the split session be terminated to the same terminal of a peer end user.
The voice call continuity, VCC, is an application provided in a home IMS network of the user, which enables a bi-directional handover of a voice call of the user between the CS domain and the IMS network. The integrated IMS architecture makes it possible to provide a popular Global System for Mobile Communications, GSM, voice call under the WLAN coverage. If the seamless voice call service is realized between the CS domain and an IP connectivity access network, IP-CAN, not only the load of GSM/UMTS radio resources is reduced, but also the gain of the operator is increased. In addition, the wired operator providing the Voice over IP, VoIP, service may also benefit from the integrated services provided by the 3GPP IMS architecture.
FIG. 2 shows an implementation architecture of 3GPP VCC. A set of functional entities are newly added in the IMS domain and the CS domain. Those functional entities are a route switch entity, a CS domain adaptation entity, a domain selection control entity, and a domain handover control entity.
(1) The route switch entity (the customized applications for mobile network enhanced logic, CAMEL, app in FIG. 2) is responsible for switching a CS domain call to the IMS domain to perform a call anchoring control. In general circumstances, the route switch entity is co-installed with the gsm service control function, SCF, in FIG. 2 and embodied as a service control point, SCP, in the CS domain.
(2) The CS domain adaptation entity (the CS adaptation function, CSAF, in FIG. 2) is responsible for receiving the CS domain call switched to the IMS domain and converting the call into an IMS domain call according to stored information (possibly interacting with the CAMEL App).
(3) The domain selection control entity (the domain selection function, DSF, in FIG. 2) is responsible for making a decision according to various policies such as a registration status and a call status of the user in the IMS domain and controlling the call to be routed to a selected connection domain.
(4) The domain handover control entity (the domain transfer function, DTF, entity in FIG. 2) is responsible for anchoring the call in the IMS domain and controlling a handover when the handover occurs.
The above four functional entities are collectively referred to as a VCC service control entity.
Based on the above VCC service control entity, when the VCC terminal is conducting a voice session of an activity, a domain handover is initiated. In order to perform the domain handover, a call initiated or accepted by the VCC terminal are all anchored to a DTF in a home IMS network of the VCC terminal. The DTF is an AS having a 3rd party call control, 3PCC, function. In the VCC, a session control leg between the DTF and the VCC terminal is called an access leg, and the session control leg between the DTF and a remote user is called a remote leg. The handover is exactly using a new access leg to replace an old access leg. In general circumstances, the VCC terminal is able to sense the strength of radio signals of access networks more accurately than a core network, and thus domain handover processes having high requirements for a delay are all initiated from the terminal towards the network. When the VCC terminal of the user detects radio signals and other factors and determines that it needs to hand over from a source network to a destination network, the VCC terminal calls a special number in the destination network. A call request for the special number may be triggered to the DTF for processing. As the original call has already been anchored at the DTF, the DTF associates the old and new calls according to a user identification, ID. A media is renegotiated with the remote user terminal of the original call according to the media in the newly established call, and a media stream corresponding to the handed-over call is switched from a port of a multi-mode terminal in the handover source network to a corresponding port in the handover destination network correspondingly during the media renegotiation process. After the media negotiation is completed, the new call is established successfully, and at this time, the call in the handover source network is released by the DTF or the VCC user equipment, UE. In this manner, the voice call of the user is handed over to the destination network. During the handover, the voice call of the user remains uninterrupted, thereby improving the user's service experience.
FIG. 3 is a flow chart of handing over a call from a CS domain to an IMS domain in the VCC.
(0-1) A VCC UE determines that the call needs to be handed over to the IMS domain according to the wireless environment and calls a special number VCC domain transfer URI, VDI, in the IMS domain to initiate a domain handover. The call request is forwarded from a proxy call session control function, P-CSCF, to a serving call session control function, S-CSCF, in the home IMS network of the VCC user for processing.
(2) The S-CSCF triggers the call to the DTF for processing according to a calling initial Filter Criteria, iFC.
(3) The DTF confirms that the call is a domain handover request according to the VDI in an INVITE, and finds the anchored session according to calling information.
(4-8) The DTF acts as an agent of the user to renegotiate a service data point (SDP) by using media information in the handover request with the remote user.
(9) After the session of the destination network is established, the DTF releases call resources in the access part of the CS domain of the VCC user.
In the process of the invention, the inventor finds that the existing VCC technology only solves the problem of handover of the voice session between bearers of different access modes, and thus can only realize the voice session continuity, but does not support the joint handover of other media sessions (for example, video, text, etc.) combinational with the voice session. That is, the conventional art cannot solve the problem of handover of a multimedia session (including several media sessions that are combinational with each other) on the multi-mode terminal between bearers of different access modes.